Replace rtc::Optional with absl::optional in test and rtc_tools
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameters 'test rtc_tools'
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: Ibb43c737f4c45fe300736382b0dd2d8ab32c6377
Reviewed-on: https://webrtc-review.googlesource.com/83944
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23642}diff --git a/rtc_tools/event_log_visualizer/analyzer.cc b/rtc_tools/event_log_visualizer/analyzer.cc
index 3bed18b..a8dbc30 100644
--- a/rtc_tools/event_log_visualizer/analyzer.cc
+++ b/rtc_tools/event_log_visualizer/analyzer.cc
@@ -130,7 +130,7 @@
// This is much more reliable for outgoing streams than for incoming streams.
template <typename RtpPacketContainer>
-rtc::Optional<uint32_t> EstimateRtpClockFrequency(
+absl::optional<uint32_t> EstimateRtpClockFrequency(
const RtpPacketContainer& packets,
int64_t end_time_us) {
RTC_CHECK(packets.size() >= 2);
@@ -151,7 +151,7 @@
<< "Failed to estimate RTP clock frequency: Stream too short. ("
<< packets.size() << " packets, "
<< last_log_timestamp - first_log_timestamp << " us)";
- return rtc::nullopt;
+ return absl::nullopt;
}
double duration =
static_cast<double>(last_log_timestamp - first_log_timestamp) /
@@ -166,7 +166,7 @@
RTC_LOG(LS_WARNING) << "Failed to estimate RTP clock frequency: Estimate "
<< estimated_frequency
<< "not close to any stardard RTP frequency.";
- return rtc::nullopt;
+ return absl::nullopt;
}
constexpr float kLeftMargin = 0.01f;
@@ -174,7 +174,7 @@
constexpr float kBottomMargin = 0.02f;
constexpr float kTopMargin = 0.05f;
-rtc::Optional<double> NetworkDelayDiff_AbsSendTime(
+absl::optional<double> NetworkDelayDiff_AbsSendTime(
const LoggedRtpPacket& old_packet,
const LoggedRtpPacket& new_packet) {
if (old_packet.header.extension.hasAbsoluteSendTime &&
@@ -188,11 +188,11 @@
recv_time_diff - AbsSendTimeToMicroseconds(send_time_diff);
return delay_change_us / 1000;
} else {
- return rtc::nullopt;
+ return absl::nullopt;
}
}
-rtc::Optional<double> NetworkDelayDiff_CaptureTime(
+absl::optional<double> NetworkDelayDiff_CaptureTime(
const LoggedRtpPacket& old_packet,
const LoggedRtpPacket& new_packet) {
int64_t send_time_diff = WrappingDifference(
@@ -230,13 +230,13 @@
// store the result in a TimeSeries.
template <typename DataType, typename IterableType>
void ProcessPoints(rtc::FunctionView<float(const DataType&)> fx,
- rtc::FunctionView<rtc::Optional<float>(const DataType&)> fy,
+ rtc::FunctionView<absl::optional<float>(const DataType&)> fy,
const IterableType& data_view,
TimeSeries* result) {
for (size_t i = 0; i < data_view.size(); i++) {
const DataType& elem = data_view[i];
float x = fx(elem);
- rtc::Optional<float> y = fy(elem);
+ absl::optional<float> y = fy(elem);
if (y)
result->points.emplace_back(x, *y);
}
@@ -248,13 +248,13 @@
template <typename DataType, typename ResultType, typename IterableType>
void ProcessPairs(
rtc::FunctionView<float(const DataType&)> fx,
- rtc::FunctionView<rtc::Optional<ResultType>(const DataType&,
- const DataType&)> fy,
+ rtc::FunctionView<absl::optional<ResultType>(const DataType&,
+ const DataType&)> fy,
const IterableType& data,
TimeSeries* result) {
for (size_t i = 1; i < data.size(); i++) {
float x = fx(data[i]);
- rtc::Optional<ResultType> y = fy(data[i - 1], data[i]);
+ absl::optional<ResultType> y = fy(data[i - 1], data[i]);
if (y)
result->points.emplace_back(x, static_cast<float>(*y));
}
@@ -266,14 +266,14 @@
template <typename DataType, typename ResultType, typename IterableType>
void AccumulatePairs(
rtc::FunctionView<float(const DataType&)> fx,
- rtc::FunctionView<rtc::Optional<ResultType>(const DataType&,
- const DataType&)> fy,
+ rtc::FunctionView<absl::optional<ResultType>(const DataType&,
+ const DataType&)> fy,
const IterableType& data,
TimeSeries* result) {
ResultType sum = 0;
for (size_t i = 1; i < data.size(); i++) {
float x = fx(data[i]);
- rtc::Optional<ResultType> y = fy(data[i - 1], data[i]);
+ absl::optional<ResultType> y = fy(data[i - 1], data[i]);
if (y)
sum += *y;
result->points.emplace_back(x, static_cast<float>(sum));
@@ -287,7 +287,7 @@
template <typename DataType, typename ResultType, typename IterableType>
void MovingAverage(
rtc::FunctionView<float(int64_t)> fx,
- rtc::FunctionView<rtc::Optional<ResultType>(const DataType&)> fy,
+ rtc::FunctionView<absl::optional<ResultType>(const DataType&)> fy,
const IterableType& data_view,
int64_t begin_time,
int64_t end_time,
@@ -301,7 +301,7 @@
for (int64_t t = begin_time; t < end_time + step; t += step) {
while (window_index_end < data_view.size() &&
data_view[window_index_end].log_time_us() < t) {
- rtc::Optional<ResultType> value = fy(data_view[window_index_end]);
+ absl::optional<ResultType> value = fy(data_view[window_index_end]);
if (value)
sum_in_window += *value;
++window_index_end;
@@ -309,7 +309,7 @@
while (window_index_begin < data_view.size() &&
data_view[window_index_begin].log_time_us() <
t - window_duration_us) {
- rtc::Optional<ResultType> value = fy(data_view[window_index_begin]);
+ absl::optional<ResultType> value = fy(data_view[window_index_begin]);
if (value)
sum_in_window -= *value;
++window_index_begin;
@@ -465,7 +465,7 @@
while (start_iter != log_start_events.end()) {
int64_t start = start_iter->log_time_us();
++start_iter;
- rtc::Optional<int64_t> next_start;
+ absl::optional<int64_t> next_start;
if (start_iter != log_start_events.end())
next_start.emplace(start_iter->log_time_us());
if (end_iter != log_end_events.end() &&
@@ -537,7 +537,7 @@
TimeSeries time_series(GetStreamName(direction, stream.ssrc),
LineStyle::kBar);
auto GetPacketSize = [](const LoggedRtpPacket& packet) {
- return rtc::Optional<float>(packet.total_length);
+ return absl::optional<float>(packet.total_length);
};
auto ToCallTime = [this](const LoggedRtpPacket& packet) {
return this->ToCallTimeSec(packet.log_time_us());
@@ -599,7 +599,7 @@
uint32_t ssrc = playout_stream.first;
if (!MatchingSsrc(ssrc, desired_ssrc_))
continue;
- rtc::Optional<int64_t> last_playout_ms;
+ absl::optional<int64_t> last_playout_ms;
TimeSeries time_series(SsrcToString(ssrc), LineStyle::kBar);
for (const auto& playout_event : playout_stream.second) {
float x = ToCallTimeSec(playout_event.log_time_us());
@@ -1139,7 +1139,7 @@
cc.OnTransportFeedback(rtcp_iterator->transport_feedback);
std::vector<PacketFeedback> feedback = cc.GetTransportFeedbackVector();
SortPacketFeedbackVector(&feedback);
- rtc::Optional<uint32_t> bitrate_bps;
+ absl::optional<uint32_t> bitrate_bps;
if (!feedback.empty()) {
#if !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
acknowledged_bitrate_estimator.IncomingPacketFeedbackVector(feedback);
@@ -1251,7 +1251,7 @@
clock.TimeInMicroseconds());
rscc.OnReceivedPacket(arrival_time_ms, payload, packet.rtp.header);
acked_bitrate.Update(payload, arrival_time_ms);
- rtc::Optional<uint32_t> bitrate_bps = acked_bitrate.Rate(arrival_time_ms);
+ absl::optional<uint32_t> bitrate_bps = acked_bitrate.Rate(arrival_time_ms);
if (bitrate_bps) {
uint32_t y = *bitrate_bps / 1000;
float x = ToCallTimeSec(clock.TimeInMicroseconds());
@@ -1383,7 +1383,7 @@
int64_t end_time_us = log_segments_.empty()
? std::numeric_limits<int64_t>::max()
: log_segments_.front().second;
- rtc::Optional<uint32_t> estimated_frequency =
+ absl::optional<uint32_t> estimated_frequency =
EstimateRtpClockFrequency(packets, end_time_us);
if (!estimated_frequency)
continue;
@@ -1463,11 +1463,11 @@
TimeSeries time_series("Audio encoder target bitrate", LineStyle::kLine,
PointStyle::kHighlight);
auto GetAnaBitrateBps = [](const LoggedAudioNetworkAdaptationEvent& ana_event)
- -> rtc::Optional<float> {
+ -> absl::optional<float> {
if (ana_event.config.bitrate_bps)
- return rtc::Optional<float>(
+ return absl::optional<float>(
static_cast<float>(*ana_event.config.bitrate_bps));
- return rtc::nullopt;
+ return absl::nullopt;
};
auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
return this->ToCallTimeSec(packet.log_time_us());
@@ -1488,9 +1488,9 @@
auto GetAnaFrameLengthMs =
[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
if (ana_event.config.frame_length_ms)
- return rtc::Optional<float>(
+ return absl::optional<float>(
static_cast<float>(*ana_event.config.frame_length_ms));
- return rtc::Optional<float>();
+ return absl::optional<float>();
};
auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
return this->ToCallTimeSec(packet.log_time_us());
@@ -1511,9 +1511,9 @@
auto GetAnaPacketLoss =
[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
if (ana_event.config.uplink_packet_loss_fraction)
- return rtc::Optional<float>(static_cast<float>(
+ return absl::optional<float>(static_cast<float>(
*ana_event.config.uplink_packet_loss_fraction));
- return rtc::Optional<float>();
+ return absl::optional<float>();
};
auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
return this->ToCallTimeSec(packet.log_time_us());
@@ -1535,9 +1535,9 @@
auto GetAnaFecEnabled =
[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
if (ana_event.config.enable_fec)
- return rtc::Optional<float>(
+ return absl::optional<float>(
static_cast<float>(*ana_event.config.enable_fec));
- return rtc::Optional<float>();
+ return absl::optional<float>();
};
auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
return this->ToCallTimeSec(packet.log_time_us());
@@ -1558,9 +1558,9 @@
auto GetAnaDtxEnabled =
[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
if (ana_event.config.enable_dtx)
- return rtc::Optional<float>(
+ return absl::optional<float>(
static_cast<float>(*ana_event.config.enable_dtx));
- return rtc::Optional<float>();
+ return absl::optional<float>();
};
auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
return this->ToCallTimeSec(packet.log_time_us());
@@ -1581,9 +1581,9 @@
auto GetAnaNumChannels =
[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
if (ana_event.config.num_channels)
- return rtc::Optional<float>(
+ return absl::optional<float>(
static_cast<float>(*ana_event.config.num_channels));
- return rtc::Optional<float>();
+ return absl::optional<float>();
};
auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
return this->ToCallTimeSec(packet.log_time_us());
@@ -1605,7 +1605,7 @@
// that outlive the one constructed.
NetEqStreamInput(const std::vector<LoggedRtpPacketIncoming>* packet_stream,
const std::vector<LoggedAudioPlayoutEvent>* output_events,
- rtc::Optional<int64_t> end_time_ms)
+ absl::optional<int64_t> end_time_ms)
: packet_stream_(*packet_stream),
packet_stream_it_(packet_stream_.begin()),
output_events_it_(output_events->begin()),
@@ -1615,22 +1615,22 @@
RTC_DCHECK(output_events);
}
- rtc::Optional<int64_t> NextPacketTime() const override {
+ absl::optional<int64_t> NextPacketTime() const override {
if (packet_stream_it_ == packet_stream_.end()) {
- return rtc::nullopt;
+ return absl::nullopt;
}
if (end_time_ms_ && packet_stream_it_->rtp.log_time_ms() > *end_time_ms_) {
- return rtc::nullopt;
+ return absl::nullopt;
}
return packet_stream_it_->rtp.log_time_ms();
}
- rtc::Optional<int64_t> NextOutputEventTime() const override {
+ absl::optional<int64_t> NextOutputEventTime() const override {
if (output_events_it_ == output_events_end_) {
- return rtc::nullopt;
+ return absl::nullopt;
}
if (end_time_ms_ && output_events_it_->log_time_ms() > *end_time_ms_) {
- return rtc::nullopt;
+ return absl::nullopt;
}
return output_events_it_->log_time_ms();
}
@@ -1661,9 +1661,9 @@
bool ended() const override { return !NextEventTime(); }
- rtc::Optional<RTPHeader> NextHeader() const override {
+ absl::optional<RTPHeader> NextHeader() const override {
if (packet_stream_it_ == packet_stream_.end()) {
- return rtc::nullopt;
+ return absl::nullopt;
}
return packet_stream_it_->rtp.header;
}
@@ -1673,7 +1673,7 @@
std::vector<LoggedRtpPacketIncoming>::const_iterator packet_stream_it_;
std::vector<LoggedAudioPlayoutEvent>::const_iterator output_events_it_;
const std::vector<LoggedAudioPlayoutEvent>::const_iterator output_events_end_;
- const rtc::Optional<int64_t> end_time_ms_;
+ const absl::optional<int64_t> end_time_ms_;
};
namespace {
@@ -1683,7 +1683,7 @@
std::unique_ptr<test::NetEqStatsGetter> CreateNetEqTestAndRun(
const std::vector<LoggedRtpPacketIncoming>* packet_stream,
const std::vector<LoggedAudioPlayoutEvent>* output_events,
- rtc::Optional<int64_t> end_time_ms,
+ absl::optional<int64_t> end_time_ms,
const std::string& replacement_file_name,
int file_sample_rate_hz) {
std::unique_ptr<test::NetEqInput> input(
@@ -1759,10 +1759,10 @@
output_events_it = parsed_log_.audio_playout_events().cbegin();
}
- rtc::Optional<int64_t> end_time_ms =
+ absl::optional<int64_t> end_time_ms =
log_segments_.empty()
- ? rtc::nullopt
- : rtc::Optional<int64_t>(log_segments_.front().second / 1000);
+ ? absl::nullopt
+ : absl::optional<int64_t>(log_segments_.front().second / 1000);
neteq_stats[ssrc] = CreateNetEqTestAndRun(
audio_packets, &output_events_it->second, end_time_ms,
@@ -1786,8 +1786,8 @@
std::vector<float> send_times_s;
std::vector<float> arrival_delay_ms;
std::vector<float> corrected_arrival_delay_ms;
- std::vector<rtc::Optional<float>> playout_delay_ms;
- std::vector<rtc::Optional<float>> target_delay_ms;
+ std::vector<absl::optional<float>> playout_delay_ms;
+ std::vector<absl::optional<float>> target_delay_ms;
neteq_stats.at(ssrc)->delay_analyzer()->CreateGraphs(
&send_times_s, &arrival_delay_ms, &corrected_arrival_delay_ms,
&playout_delay_ms, &target_delay_ms);
@@ -2014,9 +2014,9 @@
: log_segments_.front().second;
SeqNumUnwrapper<uint16_t> seq_num_unwrapper;
- rtc::Optional<int64_t> last_seq_num;
+ absl::optional<int64_t> last_seq_num;
SeqNumUnwrapper<uint32_t> capture_time_unwrapper;
- rtc::Optional<int64_t> last_capture_time;
+ absl::optional<int64_t> last_capture_time;
// Check for gaps in sequence numbers and capture timestamps.
for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
for (const auto& packet : stream.packet_view) {
@@ -2060,7 +2060,7 @@
for (const LoggedRtpPacket& rtp_packet : stream.packet_view)
rtp_in_direction.emplace(rtp_packet.log_time_us(), &rtp_packet);
}
- rtc::Optional<int64_t> last_rtp_time;
+ absl::optional<int64_t> last_rtp_time;
for (const auto& kv : rtp_in_direction) {
int64_t timestamp = kv.first;
if (timestamp > end_time_us) {
@@ -2075,7 +2075,7 @@
last_rtp_time.emplace(timestamp);
}
- rtc::Optional<int64_t> last_rtcp_time;
+ absl::optional<int64_t> last_rtcp_time;
if (direction == kIncomingPacket) {
for (const auto& rtcp : parsed_log_.incoming_rtcp_packets()) {
if (rtcp.log_time_us() > end_time_us) {